[Asterisk-Users] Re: Problem with ADIT 600 and FXO configuration
Rich Adamson
radamson at routers.com
Tue Nov 29 07:49:22 MST 2005
Couple of other items to look at... the 'zap show channels' should look
something like:
pseudo inbound-bus-lin en default
1 inbound-bus-dia en default
I don't see the 'Language' colume on your display below. Does your
zaptel.conf include:
loadzone = us
defaultzone=us
and your zapata.conf include:
[channels]
language=en
I'm using cvs-head and never have played with any of the stable versions,
so could be a difference in some of these commands and displays. Asterisk
v1.2 is very close to (if not identical) to cvs-head as of today, but
won't remain that way for very long.
What do you mean "Yes the calls out are/were to Zap/g1/xxxxxxx"?
Your outbound extensions.conf entry should look something like:
exten => _9XXXXXXX,1,Dial(Zap/g1/${EXTEN:1})
What is "xxxxxx" in your example? Copy/paste the exact entry that
you are trying to use.
Attach a voltmeter/test set across the outbound tip/ring and watch for
the needle either going to zero, or, reversing polarity about the same
time the dtmf digits are sent. Do you see anything other then maybe a
solid 10 to 15 volts? If you do see a dip to zero volts or a reversal
of polarity, then its highly likely the pstn line is using some sort of
signaling that you've not addressed as yet. (eg, maybe gound start, or
the line might have a form of "voicemail" notification on it from the
central office that was intended to blink an LED on an analog phone. I
actually used a service like that for a year or so from an Alltel
electronic central office.)
Rich
------------------------
> Hello,
> On Tue, 2005-11-29 at 02:25, Rich Adamson wrote:
> > Well... I don't have an ADIT box around, so can't help on that.
> >
> > Do take a close look at the channel assignment stuff, both in zaptel.conf
> > and zapata.conf. Are you absolutely sure the ordering of the cards
> > and channels are right (haven't moved any cards around or removed any)?
> > Your statement "it wasn't until I changed the connection to span 2 that
> > it started allowing inbound calls to work" suggests the ordering of
> > the channels might not be what you are expecting.
> >
> > You have channels 25-27 defined in zapata.conf, but they are shown as
> > unused in zaptel.conf. (I did not try to match up all the other ones.)
>
> Sorry, I had also make the requisite changes in zaptel.conf:
> span=1,0,0,esf,b8zs,yellow
> span=2,0,0,esf,b8zs
> span=3,0,0,esf,b8zs
> span=4,0,0,esf,b8zs
> fxsks=1-8
> unused=9-16
> unused=17-24
> fxsks=25-48
> unused=49-72,73-96
> fxsks=97
> fxoks=98-101
> fxoks=102-105
> loadzone = us
> defaultzone=us
>
> >
> > Take a close look at the "group=" definitions below. First set to
> > group=1, then six lines below that its group=0. Are you calling out
> > with an extensions.conf entry like Zap/g1? And, are all the channels
> > that are included in "g1" actually connected/usable? (eg, be carefull
> > with assumptions about what happens when a channel is included in the
> > group definition but the associated ADIT port isn't connected to
> > anything.) Instead of using Zap/g1, prove to yourself things are
> > configured correctly by sending calls to Zap/99 (or whichever channel
> > you have connected to a real line), and do that for each fxo line
> > that you think is wired/working.
>
> Yes the calls out are/were to Zap/g1/xxxxxxx, changing them to
> the specific Zap channels makes no difference. I just now
> tried adding "w" to the dial stream, no effect. Discovered
> that my new test-set shows DTMF digits, hooked it up and
> I'm seeing only the first digit of the phone number being
> sent on the outgoing line (the reason for the "Call didn't
> go through" message). Any ideas where next to look?
>
> > Might look at 'zap show status' and 'zap show channels' to ensure
> > what your expecting is what is defined.
>
> Is "show status" a asterisk 1.2 command?
> *CLI> zap show status
> No such command 'zap show status' (type 'help' for help)
> *CLI> zap show channels
> Chan Extension Context Language MusicOnHold
> pseudo internal default
> 25 incoming default
> 26 incoming default
> 27 incoming default
> 97 incoming default
> 98 internal default
> 99 internal default
> 100 internal default
> 101 internal default
> 102 internal default
> 103 internal default
> 104 internal default
> 105 internal default
> *CLI>
>
> > RED/NOP: RED generally means the T1 port is not seeing any timing
> > signals (eg, nothing is connected to it). NOP generally mean
> > Not-OPerational.
>
> When the cable is connected to span1 the RED goes away but it
> stays in "NOP".
>
> > Not sure why T1 port #1 on the card didn't work. Could be a bad port
> > or the channel #'s aren't as you expect. You can test for a bad port
> > by creating a T1 crossover cable, and send test calls out one T1 and
> > receive those calls on another T1 (on the same card).
>
> I may try this tomorrow, I've got about another 1/2 hour before
> I have to revert the system to original/working configuration.
>
> > Last, any changes made to zapata.conf requires a complete restart of
> > asterisk (not just a reload).
>
> That I knew and have been doing...
>
> > And, any changes to zaptel.conf requires
> > a reload of the zaptel drivers.
> >
> I thought that running ztcfg was sufficient. In any case I've
> got scripts that rmmod the modules and modprobe them before
> starting up asterisk.
>
> > Rich
> >
>
> Thanks btw. for that informative explanation of the loopstart
> v.s. groundstart signalling, could I suggest that information
> would be useful on the voip-info.org wiki (If not already there,
> I found some useful information tucked in unrelated topics).
> And I'm in the US using Qwest POTS lines so loopstart it is.
>
> > ------------------------
> >
> > > Hello,
> > > OK, some things I've found out so far. The ground connection
> > > to the ADIT chassis wasn't really to ground (fixed that, it
> > > made FXS card happy when connected).
> > >
> > > Taking a cue from another post I also reduced the number of
> > > options specified in zapata.conf to:
> > >
> > > [trunkgroups]
> > > [channels]
> > > context=default
> > > usecallerid=yes
> > > hidecallerid=no
> > > callwaiting=yes
> > > group=1
> > > callgroup=1
> > > pickupgroup=1-2
> > > immediate=no
> > > musiconhold=default
> > >
> > > group = 0
> > > signalling=fxs_ks
> > > context = incoming
> > > busydetect = no
> > > overlapdial = no
> > > channel => 25-27
> > > signalling=fxs_ks
> > > channel => 97 ;X100P
> > > group = 1
> > > signalling = fxo_ks
> > > context = internal
> > > channel => 98-100
> > > channel => 101-105
> > >
> > > Using zttool I tried to loopback the TE406P span 1 which
> > > switched the ADIT a:2 port into loop back, setting the line
> > > down and back up didn't clear the configuration (I had to
> > > find the "set a:2 line loopdown" command). Moving the link
> > > to span 2 on the TE406P I now can receive incoming calls
> > > (yea!), trying to place an outbound call results in
> > > dead air with the eventual message that the call didn't
> > > go through :-(
> > >
> > > Note that both the ADIT and the TE406P were showing
> > > green on the T1 connection however it wasn't until
> > > I changed the connection to span 2 that it started
> > > allowing inbound calls to work, zap show channel 1
> > > showed InAlarm: 1 although I didn't spot any other
> > > error messages.
> > >
> > > zztool currently shows:
> > > RED/NOP T4XXP (PCI) Card 0 Span 1
> > > OK T4XXP (PCI) Card 0 Span 2
> > > RED T4XXP (PCI) Card 0 Span 3
> > > RED T4XXP (PCI) Card 0 Span 4
> > > RED Wildcard X101P Board 1
> > > OK Wildcard TDM400P REV E/F Board 1
> > > OK Wildcard TDM400P REV E/F Board 2
> > >
> > > The "NOP" on Span 1 appears to mean "Not Opened" however
> > > I don't know what that means.
> > >
> > > I've got one more day/night to get this working so any
> > > suggestions are welcome.
> > >
> > > Thank you,
> > > William.
> > >
> > > On Mon, 2005-11-28 at 03:28, William K. Volkman wrote:
> > > > I've looked through the archives of the mailing list for the
> > > > last year and although informative I've not been successful
> > > > at get this to work. We had a working Asterisk PBX system
> > > > with 3 Digium X101P FXO lines and two TDM400P FXS cards.
> > > > I've setup an ADIT 600 with an 8 port FXO card (and an
> > > > 8 port FXS card not currently installed). We are going
> > > > to be adding a T1 for incoming calls this week. I removed
> > > > two of the X101P cards and installed a TE406P. I'm using
> > > > Asterisk 1.0.9 (and matching zaptel, libpri) from tar files.
> > > >
> > > > /etc/zaptel.conf has this configuration:
> > > > span=1,1,0,esf,b8zs,yellow
> > > > span=2,0,0,esf,b8zs
> > > > span=3,0,0,esf,b8zs
> > > > span=4,0,0,esf,b8zs
> > > > #Modular unit, first card is FXO
> > > > fxsks=1-3
> > > > unused=4-8
> > > > #Modular unit, 1 FXS cards
> > > > unused=9-16
> > > > unused=17-24
> > > > unused=25-48,49-72,73-96
> > > > fxsks=97
> > > > fxoks=98-101
> > > > fxoks=102-105
> > > >
> > > > /etc/asterisk/zapata.conf has this:
> > > > group = 0
> > > > signalling=fxs_ks
> > > > context = incoming
> > > > busydetect = yes
> > > > overlapdial = no
> > > > channel => 1-3
> > > >
> > > > signalling=fxs_ks
> > > > channel => 97 ;X100P
> > > >
> > > > group = 1
> > > > signalling = fxo_ks
> > > > context = internal
> > > > ;TDM400P
> > > > callerid = "Available" <200>
> > > > channel => 98-100
> > > > callerid = "xxxxx"
> > > > channel => 101
> > > > ;TDM400P
> > > > callerid = "xxxxx"
> > > > channel => 102
> > > > callerid = "xxxxx"
> > > > channel => 103
> > > >
> > > > Parts of my adit configuration:
> > > > -Setting slot a.
> > > >
> > > > set a:1 up
> > > > set a:1 fdl none
> > > > set a:1 lbo 4
> > > > set a:1 framing esf
> > > > set a:1 id "Inbound"
> > > > set a:1 linecode b8zs
> > > > set a:1 loopdetect csu
> > > > set a:1:1-24 side drop
> > > > set a:1:1-24 type voice
> > > > set a:1:1-24 signal ls
> > > > set a:2 up
> > > > set a:2 fdl none
> > > > set a:2 lbo 1
> > > > set a:2 framing esf
> > > > set a:2 id "Outbound PBX"
> > > > set a:2 linecode b8zs
> > > > set a:2 loopdetect csu
> > > > set a:2:1-24 side drop
> > > > set a:2:1-24 type voice
> > > > set a:2:1-24 signal ls
> > > >
> > > > -Setting slot 1.
> > > >
> > > > set 1:1-8 signal lscpd
> > > > set 1:1-8 txgain -3
> > > > set 1:1-8 rxgain -6
> > > >
> > > > -Setting primary and secondary clock sources.
> > > >
> > > > set clock1 a:1
> > > > set clock2 internal
> > > >
> > > > -Setting the system idle pattern for DS0s.
> > > >
> > > > set idle 0xff
> > > >
> > > > -Making connections.
> > > >
> > > > connect a:2:1-3 1:1-3
> > > >
> > > > Inbound calls just ring and ring (the leds on the ADIT change
> > > > state) however asterisk doesn't respond. Attempts to make
> > > > outgoing calls get:
> > > > -- Executing Dial("SIP/202-ba07", "Zap/g0/5551212") in new stack
> > > > Nov 28 02:54:45 NOTICE[8627]: app_dial.c:764 dial_exec: Unable to create
> > > > channel of type 'Zap'
> > > > == Everyone is busy/congested at this time
> > > > -- Executing Congestion("SIP/202-ba07", "") in new stack
> > > > == Spawn extension (from-sip, 95942060, 3) exited non-zero on
> > > > 'SIP/202-ba07'
> > > > -- Executing Hangup("SIP/202-ba07", "") in new stack
> > > >
> > > > I've tried just about all combinations of gs/ls/ks for the
> > > > signalling to no avail. Here is the output of status:
> > > >
> > > > > status a:2:1-3
> > > >
> > > > DS0 Rx AB Tx AB Signal T1 TP
> > > > --- ----- ----- ------ ----------------- --
> > > > a:2:1 01 01 LS Traffic N
> > > > a:2:2 01 01 LS Traffic N
> > > > a:2:3 01 01 LS Traffic N
> > > >
> > > > > status 1:1-3
> > > >
> > > > FXO Rx AB Tx AB Signal=>T1 Sig T1 TP
> > > > --- ----- ----- -------------- ----------------- --
> > > > 1:1 01 01 LSCPD => LS Traffic N
> > > > 1:2 01 01 LSCPD => LS Traffic N
> > > > 1:3 01 01 LSCPD => LS Traffic N
> > > >
> > > > > show connect a:2:1-3
> > > > From Desc Desc To
> > > > ----------- ------------------ ----------------- ---------
> > > > A:02:01 LS VOICE DS0 <--> FXO VOICE LSCPD 1:01
> > > > A:02:02 LS VOICE DS0 <--> FXO VOICE LSCPD 1:02
> > > > A:02:03 LS VOICE DS0 <--> FXO VOICE LSCPD 1:03
> > > >
> > > > Can anyone spot what I've got wrong? Any suggestions or hints
> > > > welcome.
> > > >
> > > > Thanks,
> > > > William.
>
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